Some additions: TLS/RTP instead of SRTP does also not work. There are no logs on the debug console except the message that the call is being terminated 2009-07-02 12:06:45.252177 [DEBUG] sofia.c:3100 Channel sofia/internal/835...@sip.mydomain.de entering state [terminating][0] and later cause: NORMAL_UNSPECIFIED
Best regards Peter Peter P GMX schrieb: > Hello, > > I have the following problem: Every call stops after 30 seconds when TLS > is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. > The phones are behind NAT. So I expect, that every 30 seconds an Options > request is sent. > > Wiresharking the traffic I can see > > * that there are ongoing UDP packets. > * Then a TSLv1 packet ist sent from FS to the Phone. > * This is acknowleged by the phone > * Next the phone send another UDP packet to the same FS port as before > * Then the Phone receives an ICMP request that the FS port is closed. > > > Anybody has a clue about this? > > Best regards > Peter > > > > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org